mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-06-09 17:49:26 +00:00
webrtc: update for move to gst-examples
- Integrate with the build system. - Some README updates. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-examples/-/merge_requests/16>
This commit is contained in:
Matthew Waters
parent
a88e90fa9e
commit
e1c3dad258
11
meson.build
11
meson.build
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@ -3,6 +3,12 @@ project('gst-examples', 'c', version : '1.17.1.1', license : 'LGPL')
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cc = meson.get_compiler('c')
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m_dep = cc.find_library('m', required : false)
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if cc.get_id() == 'msvc'
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add_project_arguments(
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cc.get_supported_arguments(['/utf-8']), # set the input encoding to utf-8
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language : 'c')
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endif
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glib_dep = dependency('glib-2.0', version: '>= 2.38',
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fallback: ['glib', 'libglib_dep'])
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gio_dep = dependency('gio-2.0',
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@ -18,6 +24,11 @@ gstplayer_dep = dependency('gstreamer-player-1.0', version: '>= 1.7.1.1',
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fallback: ['gst-plugins-bad', 'gstplayer_dep'])
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gsttag_dep = dependency('gstreamer-tag-1.0',
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fallback: ['gst-plugins-base', 'tag_dep'])
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gstwebrtc_dep = dependency('gstreamer-webrtc-1.0', version: '>= 1.14.0',
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fallback: ['gst-plugins-bad', 'gstwebrtc_dep'])
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gstsdp_dep = dependency('gstreamer-webrtc-1.0',
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fallback : ['gst-plugins-bad', 'gstwebrtc_dep'])
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subdir('playback')
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subdir('network')
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subdir('webrtc')
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@ -14,39 +14,35 @@ If you don't want to use the binaries provided by GStreamer or on your Linux dis
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The easiest way to build the webrtc plugin and all the plugins it needs, is to [use Cerbero](https://gstreamer.freedesktop.org/documentation/installing/building-from-source-using-cerbero.html). These instructions should work out of the box for all platforms, including cross-compiling for iOS and Android.
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One thing to note is that it's written in Python 2, so you may need to replace all instances of `./cerbero-uninstalled` (or `cerbero`) with `python2 cerbero-uninstalled` or whatever Python 2 is called on your platform.
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## Building GStreamer manually from source
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For hacking on the webrtc plugin, you may want to build manually using the git repositories:
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- http://gitlab.freedesktop.org/gstreamer/gstreamer
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- http://gitlab.freedesktop.org/gstreamer/gst-plugins-base
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- http://gitlab.freedesktop.org/gstreamer/gst-plugins-good
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- http://gitlab.freedesktop.org/gstreamer/gst-plugins-bad
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- http://gitlab.freedesktop.org/libnice/libnice
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Or with Meson gst-build:
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https://gitlab.freedesktop.org/gstreamer/gst-build/
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You may need to install the following packages using your package manager:
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json-glib, libsoup, libnice, libnice-gstreamer1 (the gstreamer plugin for libnice, called gstreamer1.0-nice Debian)
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### Ubuntu 18.04
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Here are the commands for Ubuntu 18.04.
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```
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sudo apt-get install -y gstreamer1.0-tools gstreamer1.0-nice gstreamer1.0-plugins-bad gstreamer1.0-plugins-ugly gstreamer1.0-plugins-good libgstreamer1.0-dev git libglib2.0-dev libgstreamer-plugins-bad1.0-dev libsoup2.4-dev libjson-glib-dev
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```
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For hacking on the webrtc plugin, you may want to build manually using the git repositories:
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- http://cgit.freedesktop.org/gstreamer/gstreamer
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- http://cgit.freedesktop.org/gstreamer/gst-plugins-base
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- http://cgit.freedesktop.org/gstreamer/gst-plugins-good
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- http://cgit.freedesktop.org/gstreamer/gst-plugins-bad
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- http://cgit.freedesktop.org/libnice/libnice
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You can build these with either Autotools gst-uninstalled:
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https://arunraghavan.net/2014/07/quick-start-guide-to-gst-uninstalled-1-x/
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Or with Meson gst-build:
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https://cgit.freedesktop.org/gstreamer/gst-build/
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You may need to install the following packages using your package manager:
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json-glib, libsoup, libnice, libnice-gstreamer1 (the gstreamer plugin for libnice, called gstreamer1.0-nice Debian)
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## Filing bugs
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Please only file bugs about the demos here. Bugs about GStreamer's WebRTC implementation should be filed on the [GStreamer bugzilla](https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-plugins-bad).
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Please only file bugs about the demos here. Bugs about GStreamer's WebRTC implementation should be filed on the [GStreamer gitlab](https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/issues/new).
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You can also find us on IRC by joining #gstreamer @ FreeNode.
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@ -58,6 +54,18 @@ http://blog.nirbheek.in/2018/02/gstreamer-webrtc.html
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## Examples
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### Building
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Most of the examples that require a build process can be built using the meson build system in the top-level gst-examples directory by using the following commands:
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```console
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cd /path/to/gst-examples
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meson _builddir
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ninja -C _builddir
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```
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Build outputs will be placed in the directory `_builddir`.
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### sendrecv: Send and receive audio and video
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* Serve the `js/` directory on the root of your website, or open https://webrtc.nirbheek.in
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@ -67,12 +75,6 @@ http://blog.nirbheek.in/2018/02/gstreamer-webrtc.html
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#### Running the C version
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* Build the sources in the `gst/` directory on your machine. Use `make` or
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```console
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$ gcc webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o webrtc-sendrecv
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```
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* Run `webrtc-sendrecv --peer-id=ID` with the `id` from the browser. You will see state changes and an SDP exchange.
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#### Running the Python version
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@ -104,13 +106,7 @@ You can optionally specify the server URL too (it defaults to wss://webrtc.nirbh
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### multiparty-sendrecv: Multiparty audio conference with N peers
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* Build the sources in the `gst/` directory on your machine
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```console
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$ gcc mp-webrtc-sendrecv.c $(pkg-config --cflags --libs gstreamer-webrtc-1.0 gstreamer-sdp-1.0 libsoup-2.4 json-glib-1.0) -o mp-webrtc-sendrecv
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```
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* Run `mp-webrtc-sendrecv --room-id=ID` with `ID` as a room name. The peer will connect to the signalling server and setup a conference room.
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* Run `_builddir/multiparty-sendrecv/gst/mp-webrtc-sendrecv --room-id=ID` with `ID` as a room name. The peer will connect to the signalling server and setup a conference room.
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* Run this as many times as you like, each will spawn a peer that sends red noise and outputs the red noise it receives from other peers.
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- To change what a peer sends, find the `audiotestsrc` element in the source and change the `wave` property.
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- You can, of course, also replace `audiotestsrc` itself with `autoaudiosrc` (any platform) or `pulsesink` (on linux).
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@ -2,14 +2,17 @@ tests = [
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['basic', 'basic.py'],
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]
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test_deps = [certs]
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if openssl.found()
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test_deps = [certs]
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foreach elem : tests
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test(elem.get(0),
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py3,
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depends: test_deps,
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args : files(elem.get(1)),
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env : ['PYTHONPATH=' + join_paths(meson.source_root(), 'sendrecv', 'gst') + ':' + join_paths(meson.source_root(), 'signalling'),
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'TEST_HTML_SOURCE=' + join_paths(meson.source_root(), 'sendrecv', 'js'),
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'TEST_CA_CERT_PATH=' + join_paths(meson.build_root(), 'signalling')])
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endforeach
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foreach elem : tests
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test(elem.get(0),
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py3,
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depends: test_deps,
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args : files(elem.get(1)),
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# XXX: This PYTHONPATH hack is fun
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env : ['PYTHONPATH=' + join_paths(meson.source_root(), 'sendrecv', 'gst') + ':' + join_paths(meson.source_root(), 'signalling'),
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'TEST_HTML_SOURCE=' + join_paths(meson.source_root(), 'sendrecv', 'js'),
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'TEST_CA_CERT_PATH=' + join_paths(meson.build_root(), 'signalling')])
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endforeach
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endif
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@ -1,31 +1,12 @@
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project('gstwebrtc-demo', 'c',
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meson_version : '>= 0.48',
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license: 'BSD-2-Clause',
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default_options : [ 'warning_level=1',
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'buildtype=debug' ])
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cc = meson.get_compiler('c')
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if cc.get_id() == 'msvc'
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add_project_arguments(
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cc.get_supported_arguments(['/utf-8']), # set the input encoding to utf-8
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language : 'c')
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endif
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gst_req = '>= 1.14.0'
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gst_dep = dependency('gstreamer-1.0', version : gst_req,
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fallback : ['gstreamer', 'gst_dep'])
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gstsdp_dep = dependency('gstreamer-sdp-1.0', version : gst_req,
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fallback : ['gst-plugins-base', 'sdp_dep'])
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gstwebrtc_dep = dependency('gstreamer-webrtc-1.0', version : gst_req,
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fallback : ['gst-plugins-bad', 'gstwebrtc_dep'])
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libsoup_dep = dependency('libsoup-2.4', version : '>=2.48',
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fallback : ['libsoup', 'libsoup_dep'])
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json_glib_dep = dependency('json-glib-1.0',
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fallback : ['json-glib', 'json_glib_dep'])
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py3_mod = import('python3')
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py3 = py3_mod.find_python()
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@ -7,4 +7,10 @@ if [ $# -eq 1 ]; then
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OUTDIR=$1/
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fi
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openssl req -x509 -newkey rsa:4096 -keyout ${OUTDIR}key.pem -out ${OUTDIR}cert.pem -days 365 -nodes -subj "/CN=example.com"
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output=$(openssl req -x509 -newkey rsa:4096 -keyout ${OUTDIR}key.pem -out ${OUTDIR}cert.pem -days 365 -nodes -subj "/CN=example.com" 2>&1)
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ret=$?
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if [ ! $ret -eq 0 ]; then
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echo "${output}" 1>&2
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exit $ret
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fi
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@ -1,8 +1,12 @@
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generate_certs = find_program('generate_cert.sh')
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certs = custom_target(
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simple_server = files('simple_server.py')
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openssl = find_program ('openssl', required : false)
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if openssl.found()
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# TODO: a better cert generation algorithm that doesn't rely on calling openssl
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generate_certs = find_program('generate_cert.sh')
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certs = custom_target(
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'generate-certs',
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command: [generate_certs, '@OUTDIR@'],
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output : ['key.pem', 'cert.pem']
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)
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simple_server = files('simple_server.py')
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)
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endif
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